Add algorithm for Residual Echo Detector.

This algorithm calculates an estimate of the Pearson product-moment correlation coefficient between the power of 10ms audio buffers taken from the render and capture sides, for various different delay values.

BUG=webrtc:6525

Review-Url: https://codereview.webrtc.org/2419563003
Cr-Commit-Position: refs/heads/master@{#14824}

webrtc/modules/BUILD.gn

@@ -589,11 +589,15 @@ if (rtc_include_tests) {
         "audio_processing/audio_processing_unittest.cc",
         "audio_processing/echo_cancellation_bit_exact_unittest.cc",
         "audio_processing/echo_control_mobile_unittest.cc",
+        "audio_processing/echo_detector/circular_buffer_unittest.cc",
+        "audio_processing/echo_detector/mean_variance_estimator_unittest.cc",
+        "audio_processing/echo_detector/normalized_covariance_estimator_unittest.cc",
         "audio_processing/gain_control_unittest.cc",
         "audio_processing/high_pass_filter_unittest.cc",
         "audio_processing/level_controller/level_controller_unittest.cc",
         "audio_processing/level_estimator_unittest.cc",
         "audio_processing/noise_suppression_unittest.cc",
+        "audio_processing/residual_echo_detector_unittest.cc",
         "audio_processing/test/bitexactness_tools.cc",
         "audio_processing/test/bitexactness_tools.h",
         "audio_processing/test/debug_dump_replayer.cc",

webrtc/modules/audio_processing/BUILD.gn

@@ -61,6 +61,12 @@ rtc_static_library("audio_processing") {
     "echo_cancellation_impl.h",
     "echo_control_mobile_impl.cc",
     "echo_control_mobile_impl.h",
+    "echo_detector/circular_buffer.cc",
+    "echo_detector/circular_buffer.h",
+    "echo_detector/mean_variance_estimator.cc",
+    "echo_detector/mean_variance_estimator.h",
+    "echo_detector/normalized_covariance_estimator.cc",
+    "echo_detector/normalized_covariance_estimator.h",
     "gain_control_for_experimental_agc.cc",
     "gain_control_for_experimental_agc.h",
     "gain_control_impl.cc",

webrtc/modules/audio_processing/audio_processing.gypi

@@ -73,6 +73,12 @@
         'echo_cancellation_impl.h',
         'echo_control_mobile_impl.cc',
         'echo_control_mobile_impl.h',
+        'echo_detector/circular_buffer.cc',
+        'echo_detector/circular_buffer.h',
+        'echo_detector/mean_variance_estimator.cc',
+        'echo_detector/mean_variance_estimator.h',
+        'echo_detector/normalized_covariance_estimator.cc',
+        'echo_detector/normalized_covariance_estimator.h',
         'gain_control_for_experimental_agc.cc',
         'gain_control_for_experimental_agc.h',
         'gain_control_impl.cc',

webrtc/modules/audio_processing/echo_detector/circular_buffer.cc

@@ -0,0 +1,49 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "webrtc/modules/audio_processing/echo_detector/circular_buffer.h"
+
+#include <algorithm>
+
+#include "webrtc/base/checks.h"
+
+namespace webrtc {
+
+CircularBuffer::CircularBuffer(size_t size) : buffer_(size) {}
+CircularBuffer::~CircularBuffer() = default;
+
+void CircularBuffer::Push(float value) {
+  buffer_[next_insertion_index_] = value;
+  ++next_insertion_index_;
+  next_insertion_index_ %= buffer_.size();
+  RTC_DCHECK_LT(next_insertion_index_, buffer_.size());
+  nr_elements_in_buffer_ = std::min(nr_elements_in_buffer_ + 1, buffer_.size());
+  RTC_DCHECK_LE(nr_elements_in_buffer_, buffer_.size());
+}
+
+rtc::Optional<float> CircularBuffer::Pop() {
+  if (nr_elements_in_buffer_ == 0) {
+    return rtc::Optional<float>();
+  }
+  const size_t index =
+      (buffer_.size() + next_insertion_index_ - nr_elements_in_buffer_) %
+      buffer_.size();
+  RTC_DCHECK_LT(index, buffer_.size());
+  --nr_elements_in_buffer_;
+  return rtc::Optional<float>(buffer_[index]);
+}
+
+void CircularBuffer::Clear() {
+  std::fill(buffer_.begin(), buffer_.end(), 0.f);
+  next_insertion_index_ = 0;
+  nr_elements_in_buffer_ = 0;
+}
+
+}  // namespace webrtc

webrtc/modules/audio_processing/echo_detector/circular_buffer.h

@@ -0,0 +1,42 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_ECHO_DETECTOR_CIRCULAR_BUFFER_H_
+#define WEBRTC_MODULES_AUDIO_PROCESSING_ECHO_DETECTOR_CIRCULAR_BUFFER_H_
+
+#include <vector>
+
+#include "webrtc/base/optional.h"
+
+namespace webrtc {
+
+// Ring buffer containing floating point values.
+struct CircularBuffer {
+ public:
+  explicit CircularBuffer(size_t size);
+  ~CircularBuffer();
+
+  void Push(float value);
+  rtc::Optional<float> Pop();
+  size_t Size() const { return nr_elements_in_buffer_; }
+  // This function fills the buffer with zeros, but does not change its size.
+  void Clear();
+
+ private:
+  std::vector<float> buffer_;
+  size_t next_insertion_index_ = 0;
+  // This is the number of elements that have been pushed into the circular
+  // buffer, not the allocated buffer size.
+  size_t nr_elements_in_buffer_ = 0;
+};
+
+}  // namespace webrtc
+
+#endif  // WEBRTC_MODULES_AUDIO_PROCESSING_ECHO_DETECTOR_CIRCULAR_BUFFER_H_

webrtc/modules/audio_processing/echo_detector/circular_buffer_unittest.cc

@@ -0,0 +1,52 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "webrtc/modules/audio_processing/echo_detector/circular_buffer.h"
+#include "webrtc/test/gtest.h"
+
+namespace webrtc {
+
+TEST(CircularBufferTests, LessThanMaxTest) {
+  CircularBuffer test_buffer(3);
+  test_buffer.Push(1.f);
+  test_buffer.Push(2.f);
+  EXPECT_EQ(rtc::Optional<float>(1.f), test_buffer.Pop());
+  EXPECT_EQ(rtc::Optional<float>(2.f), test_buffer.Pop());
+}
+
+TEST(CircularBufferTests, FillTest) {
+  CircularBuffer test_buffer(3);
+  test_buffer.Push(1.f);
+  test_buffer.Push(2.f);
+  test_buffer.Push(3.f);
+  EXPECT_EQ(rtc::Optional<float>(1.f), test_buffer.Pop());
+  EXPECT_EQ(rtc::Optional<float>(2.f), test_buffer.Pop());
+  EXPECT_EQ(rtc::Optional<float>(3.f), test_buffer.Pop());
+}
+
+TEST(CircularBufferTests, OverflowTest) {
+  CircularBuffer test_buffer(3);
+  test_buffer.Push(1.f);
+  test_buffer.Push(2.f);
+  test_buffer.Push(3.f);
+  test_buffer.Push(4.f);
+  // Because the circular buffer has a size of 3, the first insert should have
+  // been forgotten.
+  EXPECT_EQ(rtc::Optional<float>(2.f), test_buffer.Pop());
+  EXPECT_EQ(rtc::Optional<float>(3.f), test_buffer.Pop());
+  EXPECT_EQ(rtc::Optional<float>(4.f), test_buffer.Pop());
+}
+
+TEST(CircularBufferTests, ReadFromEmpty) {
+  CircularBuffer test_buffer(3);
+  EXPECT_EQ(rtc::Optional<float>(), test_buffer.Pop());
+}
+
+}  // namespace webrtc

webrtc/modules/audio_processing/echo_detector/mean_variance_estimator.cc

@@ -0,0 +1,45 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "webrtc/modules/audio_processing/echo_detector/mean_variance_estimator.h"
+
+#include <math.h>
+
+#include "webrtc/base/checks.h"
+
+namespace webrtc {
+namespace {
+
+// Parameter controlling the adaptation speed.
+constexpr float kAlpha = 0.01f;
+
+}  // namespace
+
+void MeanVarianceEstimator::Update(float value) {
+  mean_ = (1.f - kAlpha) * mean_ + kAlpha * value;
+  variance_ =
+      (1.f - kAlpha) * variance_ + kAlpha * (value - mean_) * (value - mean_);
+}
+
+float MeanVarianceEstimator::std_deviation() const {
+  RTC_DCHECK_GE(variance_, 0.f);
+  return sqrtf(variance_);
+}
+
+float MeanVarianceEstimator::mean() const {
+  return mean_;
+}
+
+void MeanVarianceEstimator::Clear() {
+  mean_ = 0.f;
+  variance_ = 0.f;
+}
+
+}  // namespace webrtc

webrtc/modules/audio_processing/echo_detector/mean_variance_estimator.h

@@ -0,0 +1,33 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_ECHO_DETECTOR_MEAN_VARIANCE_ESTIMATOR_H_
+#define WEBRTC_MODULES_AUDIO_PROCESSING_ECHO_DETECTOR_MEAN_VARIANCE_ESTIMATOR_H_
+
+namespace webrtc {
+
+// This class iteratively estimates the mean and variance of a signal.
+class MeanVarianceEstimator {
+ public:
+  void Update(float value);
+  float std_deviation() const;
+  float mean() const;
+  void Clear();
+
+ private:
+  // Estimate of the expected value of the input values.
+  float mean_ = 0.f;
+  // Estimate of the variance of the input values.
+  float variance_ = 0.f;
+};
+
+}  // namespace webrtc
+
+#endif  // WEBRTC_MODULES_AUDIO_PROCESSING_ECHO_DETECTOR_MEAN_VARIANCE_ESTIMATOR_H_

webrtc/modules/audio_processing/echo_detector/mean_variance_estimator_unittest.cc

@@ -0,0 +1,64 @@
+
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "webrtc/modules/audio_processing/echo_detector/mean_variance_estimator.h"
+#include "webrtc/test/gtest.h"
+
+namespace webrtc {
+
+TEST(MeanVarianceEstimatorTests, InsertTwoValues) {
+  MeanVarianceEstimator test_estimator;
+  // Insert two values.
+  test_estimator.Update(3.f);
+  test_estimator.Update(5.f);
+
+  EXPECT_GT(test_estimator.mean(), 0.f);
+  EXPECT_GT(test_estimator.std_deviation(), 0.f);
+  // Test Clear method
+  test_estimator.Clear();
+  EXPECT_EQ(test_estimator.mean(), 0.f);
+  EXPECT_EQ(test_estimator.std_deviation(), 0.f);
+}
+
+TEST(MeanVarianceEstimatorTests, InsertZeroes) {
+  MeanVarianceEstimator test_estimator;
+  // Insert the same value many times.
+  for (size_t i = 0; i < 10000; i++) {
+    test_estimator.Update(0.f);
+  }
+  EXPECT_EQ(test_estimator.mean(), 0.f);
+  EXPECT_EQ(test_estimator.std_deviation(), 0.f);
+}
+
+TEST(MeanVarianceEstimatorTests, ConstantValueTest) {
+  MeanVarianceEstimator test_estimator;
+  for (size_t i = 0; i < 10000; i++) {
+    test_estimator.Update(3.f);
+  }
+  // The mean should be close to three, and the standard deviation should be
+  // close to zero.
+  EXPECT_NEAR(3.0f, test_estimator.mean(), 0.01f);
+  EXPECT_NEAR(0.0f, test_estimator.std_deviation(), 0.01f);
+}
+
+TEST(MeanVarianceEstimatorTests, AlternatingValueTest) {
+  MeanVarianceEstimator test_estimator;
+  for (size_t i = 0; i < 10000; i++) {
+    test_estimator.Update(1.f);
+    test_estimator.Update(-1.f);
+  }
+  // The mean should be close to zero, and the standard deviation should be
+  // close to one.
+  EXPECT_NEAR(0.0f, test_estimator.mean(), 0.01f);
+  EXPECT_NEAR(1.0f, test_estimator.std_deviation(), 0.01f);
+}
+
+}  // namespace webrtc

webrtc/modules/audio_processing/echo_detector/normalized_covariance_estimator.cc

@@ -0,0 +1,37 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "webrtc/modules/audio_processing/echo_detector/normalized_covariance_estimator.h"
+
+namespace webrtc {
+namespace {
+
+// Parameter controlling the adaptation speed.
+constexpr float kAlpha = 0.01f;
+
+}  // namespace
+
+void NormalizedCovarianceEstimator::Update(float x,
+                                           float x_mean,
+                                           float x_sigma,
+                                           float y,
+                                           float y_mean,
+                                           float y_sigma) {
+  covariance_ =
+      (1.f - kAlpha) * covariance_ + kAlpha * (x - x_mean) * (y - y_mean);
+  normalized_cross_correlation_ = covariance_ / (x_sigma * y_sigma + .0001f);
+}
+
+void NormalizedCovarianceEstimator::Clear() {
+  covariance_ = 0.f;
+  normalized_cross_correlation_ = 0.f;
+}
+
+}  // namespace webrtc

webrtc/modules/audio_processing/echo_detector/normalized_covariance_estimator.h

@@ -0,0 +1,42 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_ECHO_DETECTOR_NORMALIZED_COVARIANCE_ESTIMATOR_H_
+#define WEBRTC_MODULES_AUDIO_PROCESSING_ECHO_DETECTOR_NORMALIZED_COVARIANCE_ESTIMATOR_H_
+
+namespace webrtc {
+
+// This class iteratively estimates the normalized covariance between two
+// signals.
+class NormalizedCovarianceEstimator {
+ public:
+  void Update(float x,
+              float x_mean,
+              float x_var,
+              float y,
+              float y_mean,
+              float y_var);
+  // This function returns an estimate of the Pearson product-moment correlation
+  // coefficient of the two signals.
+  float normalized_cross_correlation() const {
+    return normalized_cross_correlation_;
+  }
+  // This function resets the estimated values to zero.
+  void Clear();
+
+ private:
+  float normalized_cross_correlation_ = 0.f;
+  // Estimate of the covariance value.
+  float covariance_ = 0.f;
+};
+
+}  // namespace webrtc
+
+#endif  // WEBRTC_MODULES_AUDIO_PROCESSING_ECHO_DETECTOR_NORMALIZED_COVARIANCE_ESTIMATOR_H_

webrtc/modules/audio_processing/echo_detector/normalized_covariance_estimator_unittest.cc

@@ -0,0 +1,40 @@
+
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "webrtc/modules/audio_processing/echo_detector/normalized_covariance_estimator.h"
+#include "webrtc/test/gtest.h"
+
+namespace webrtc {
+
+TEST(NormalizedCovarianceEstimatorTests, IdenticalSignalTest) {
+  NormalizedCovarianceEstimator test_estimator;
+  for (size_t i = 0; i < 10000; i++) {
+    test_estimator.Update(1.f, 0.f, 1.f, 1.f, 0.f, 1.f);
+    test_estimator.Update(-1.f, 0.f, 1.f, -1.f, 0.f, 1.f);
+  }
+  // A normalized covariance value close to 1 is expected.
+  EXPECT_NEAR(1.f, test_estimator.normalized_cross_correlation(), 0.01f);
+  test_estimator.Clear();
+  EXPECT_EQ(0.f, test_estimator.normalized_cross_correlation());
+}
+
+TEST(NormalizedCovarianceEstimatorTests, OppositeSignalTest) {
+  NormalizedCovarianceEstimator test_estimator;
+  // Insert the same value many times.
+  for (size_t i = 0; i < 10000; i++) {
+    test_estimator.Update(1.f, 0.f, 1.f, -1.f, 0.f, 1.f);
+    test_estimator.Update(-1.f, 0.f, 1.f, 1.f, 0.f, 1.f);
+  }
+  // A normalized covariance value close to -1 is expected.
+  EXPECT_NEAR(-1.f, test_estimator.normalized_cross_correlation(), 0.01f);
+}
+
+}  // namespace webrtc

webrtc/modules/audio_processing/residual_echo_detector.cc

@@ -10,26 +10,113 @@
 
 #include "webrtc/modules/audio_processing/residual_echo_detector.h"
 
+#include <algorithm>
+#include <numeric>
+
 #include "webrtc/modules/audio_processing/audio_buffer.h"
 
+namespace {
+
+float Power(rtc::ArrayView<const float> input) {
+  return std::inner_product(input.begin(), input.end(), input.begin(), 0.f);
+}
+
+constexpr size_t kLookbackFrames = 650;
+// TODO(ivoc): Verify the size of this buffer.
+constexpr size_t kRenderBufferSize = 30;
+
+}  // namespace
+
 namespace webrtc {
 
-ResidualEchoDetector::ResidualEchoDetector() = default;
+ResidualEchoDetector::ResidualEchoDetector()
+    : render_buffer_(kRenderBufferSize),
+      render_power_(kLookbackFrames),
+      render_power_mean_(kLookbackFrames),
+      render_power_std_dev_(kLookbackFrames),
+      covariances_(kLookbackFrames){};
 
 ResidualEchoDetector::~ResidualEchoDetector() = default;
 
 void ResidualEchoDetector::AnalyzeRenderAudio(
-    rtc::ArrayView<const float> render_audio) const {
-  // TODO(ivoc): Add implementation.
+    rtc::ArrayView<const float> render_audio) {
+  if (render_buffer_.Size() == 0) {
+    frames_since_zero_buffer_size_ = 0;
+  } else if (frames_since_zero_buffer_size_ >= kRenderBufferSize) {
+    // This can happen in a few cases: at the start of a call, due to a glitch
+    // or due to clock drift. The excess capture value will be ignored.
+    // TODO(ivoc): Include how often this happens in APM stats.
+    render_buffer_.Pop();
+    frames_since_zero_buffer_size_ = 0;
+  }
+  ++frames_since_zero_buffer_size_;
+  float power = Power(render_audio);
+  render_buffer_.Push(power);
 }
 
 void ResidualEchoDetector::AnalyzeCaptureAudio(
-    rtc::ArrayView<const float> render_audio) {
-  // TODO(ivoc): Add implementation.
+    rtc::ArrayView<const float> capture_audio) {
+  if (first_process_call_) {
+    // On the first process call (so the start of a call), we must flush the
+    // render buffer, otherwise the render data will be delayed.
+    render_buffer_.Clear();
+    first_process_call_ = false;
+  }
+
+  // Get the next render value.
+  const rtc::Optional<float> buffered_render_power = render_buffer_.Pop();
+  if (!buffered_render_power) {
+    // This can happen in a few cases: at the start of a call, due to a glitch
+    // or due to clock drift. The excess capture value will be ignored.
+    // TODO(ivoc): Include how often this happens in APM stats.
+    return;
+  }
+  // Update the render statistics, and store the statistics in circular buffers.
+  render_statistics_.Update(*buffered_render_power);
+  RTC_DCHECK_LT(next_insertion_index_, kLookbackFrames);
+  render_power_[next_insertion_index_] = *buffered_render_power;
+  render_power_mean_[next_insertion_index_] = render_statistics_.mean();
+  render_power_std_dev_[next_insertion_index_] =
+      render_statistics_.std_deviation();
+
+  // Get the next capture value, update capture statistics and add the relevant
+  // values to the buffers.
+  const float capture_power = Power(capture_audio);
+  capture_statistics_.Update(capture_power);
+  const float capture_mean = capture_statistics_.mean();
+  const float capture_std_deviation = capture_statistics_.std_deviation();
+
+  // Update the covariance values and determine the new echo likelihood.
+  echo_likelihood_ = 0.f;
+  for (size_t delay = 0; delay < covariances_.size(); ++delay) {
+    const size_t read_index =
+        (kLookbackFrames + next_insertion_index_ - delay) % kLookbackFrames;
+    RTC_DCHECK_LT(read_index, render_power_.size());
+    covariances_[delay].Update(capture_power, capture_mean,
+                               capture_std_deviation, render_power_[read_index],
+                               render_power_mean_[read_index],
+                               render_power_std_dev_[read_index]);
+    echo_likelihood_ = std::max(
+        echo_likelihood_, covariances_[delay].normalized_cross_correlation());
+  }
+
+  // Update the next insertion index.
+  ++next_insertion_index_;
+  next_insertion_index_ %= kLookbackFrames;
 }
 
 void ResidualEchoDetector::Initialize() {
-  // TODO(ivoc): Add implementation.
+  render_buffer_.Clear();
+  std::fill(render_power_.begin(), render_power_.end(), 0.f);
+  std::fill(render_power_mean_.begin(), render_power_mean_.end(), 0.f);
+  std::fill(render_power_std_dev_.begin(), render_power_std_dev_.end(), 0.f);
+  render_statistics_.Clear();
+  capture_statistics_.Clear();
+  for (auto& cov : covariances_) {
+    cov.Clear();
+  }
+  echo_likelihood_ = 0.f;
+  next_insertion_index_ = 0;
 }
 
 void ResidualEchoDetector::PackRenderAudioBuffer(
@@ -44,9 +131,4 @@ void ResidualEchoDetector::PackRenderAudioBuffer(
                          audio->num_frames_per_band()));
 }
 
-float ResidualEchoDetector::get_echo_likelihood() const {
-  // TODO(ivoc): Add implementation.
-  return 0.0f;
-}
-
 }  // namespace webrtc

webrtc/modules/audio_processing/residual_echo_detector.h

@@ -11,10 +11,12 @@
 #ifndef WEBRTC_MODULES_AUDIO_PROCESSING_RESIDUAL_ECHO_DETECTOR_H_
 #define WEBRTC_MODULES_AUDIO_PROCESSING_RESIDUAL_ECHO_DETECTOR_H_
 
-#include <memory>
 #include <vector>
 
 #include "webrtc/base/array_view.h"
+#include "webrtc/modules/audio_processing/echo_detector/circular_buffer.h"
+#include "webrtc/modules/audio_processing/echo_detector/mean_variance_estimator.h"
+#include "webrtc/modules/audio_processing/echo_detector/normalized_covariance_estimator.h"
 
 namespace webrtc {
 
@@ -27,7 +29,7 @@ class ResidualEchoDetector {
   ~ResidualEchoDetector();
 
   // This function should be called while holding the render lock.
-  void AnalyzeRenderAudio(rtc::ArrayView<const float> render_audio) const;
+  void AnalyzeRenderAudio(rtc::ArrayView<const float> render_audio);
 
   // This function should be called while holding the capture lock.
   void AnalyzeCaptureAudio(rtc::ArrayView<const float> capture_audio);
@@ -39,7 +41,36 @@ class ResidualEchoDetector {
                                     std::vector<float>* packed_buffer);
 
   // This function should be called while holding the capture lock.
-  float get_echo_likelihood() const;
+  float echo_likelihood() const { return echo_likelihood_; }
+
+ private:
+  // Keep track if the |Process| function has been previously called.
+  bool first_process_call_ = true;
+  // Buffer for storing the power of incoming farend buffers. This is needed for
+  // cases where calls to BufferFarend and Process are jittery.
+  CircularBuffer render_buffer_;
+  // Count how long ago it was that the size of |render_buffer_| was zero. This
+  // value is also reset to zero when clock drift is detected and a value from
+  // the renderbuffer is discarded, even though the buffer is not actually zero
+  // at that point. This is done to avoid repeatedly removing elements in this
+  // situation.
+  size_t frames_since_zero_buffer_size_ = 0;
+
+  // Circular buffers containing delayed versions of the power, mean and
+  // standard deviation, for calculating the delayed covariance values.
+  std::vector<float> render_power_;
+  std::vector<float> render_power_mean_;
+  std::vector<float> render_power_std_dev_;
+  // Covariance estimates for different delay values.
+  std::vector<NormalizedCovarianceEstimator> covariances_;
+  // Index where next element should be inserted in all of the above circular
+  // buffers.
+  size_t next_insertion_index_ = 0;
+
+  MeanVarianceEstimator render_statistics_;
+  MeanVarianceEstimator capture_statistics_;
+  // Current echo likelihood.
+  float echo_likelihood_ = 0.f;
 };
 
 }  // namespace webrtc

webrtc/modules/audio_processing/residual_echo_detector_unittest.cc

@@ -0,0 +1,124 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include <vector>
+
+#include "webrtc/modules/audio_processing/residual_echo_detector.h"
+#include "webrtc/test/gtest.h"
+
+namespace webrtc {
+
+TEST(ResidualEchoDetectorTests, Echo) {
+  ResidualEchoDetector echo_detector;
+  std::vector<float> ones(160, 1.f);
+  std::vector<float> zeros(160, 0.f);
+
+  // In this test the capture signal has a delay of 10 frames w.r.t. the render
+  // signal, but is otherwise identical. Both signals are periodic with a 20
+  // frame interval.
+  for (int i = 0; i < 1000; i++) {
+    if (i % 20 == 0) {
+      echo_detector.AnalyzeRenderAudio(ones);
+      echo_detector.AnalyzeCaptureAudio(zeros);
+    } else if (i % 20 == 10) {
+      echo_detector.AnalyzeRenderAudio(zeros);
+      echo_detector.AnalyzeCaptureAudio(ones);
+    } else {
+      echo_detector.AnalyzeRenderAudio(zeros);
+      echo_detector.AnalyzeCaptureAudio(zeros);
+    }
+  }
+  // We expect to detect echo with near certain likelihood.
+  EXPECT_NEAR(1.f, echo_detector.echo_likelihood(), 0.01f);
+}
+
+TEST(ResidualEchoDetectorTests, NoEcho) {
+  ResidualEchoDetector echo_detector;
+  std::vector<float> ones(160, 1.f);
+  std::vector<float> zeros(160, 0.f);
+
+  // In this test the capture signal is always zero, so no echo should be
+  // detected.
+  for (int i = 0; i < 1000; i++) {
+    if (i % 20 == 0) {
+      echo_detector.AnalyzeRenderAudio(ones);
+    } else {
+      echo_detector.AnalyzeRenderAudio(zeros);
+    }
+    echo_detector.AnalyzeCaptureAudio(zeros);
+  }
+  // We expect to not detect any echo.
+  EXPECT_NEAR(0.f, echo_detector.echo_likelihood(), 0.01f);
+}
+
+TEST(ResidualEchoDetectorTests, EchoWithRenderClockDrift) {
+  ResidualEchoDetector echo_detector;
+  std::vector<float> ones(160, 1.f);
+  std::vector<float> zeros(160, 0.f);
+
+  // In this test the capture signal has a delay of 10 frames w.r.t. the render
+  // signal, but is otherwise identical. Both signals are periodic with a 20
+  // frame interval. There is a simulated clock drift of 1% in this test, with
+  // the render side producing data slightly faster.
+  for (int i = 0; i < 1000; i++) {
+    if (i % 20 == 0) {
+      echo_detector.AnalyzeRenderAudio(ones);
+      echo_detector.AnalyzeCaptureAudio(zeros);
+    } else if (i % 20 == 10) {
+      echo_detector.AnalyzeRenderAudio(zeros);
+      echo_detector.AnalyzeCaptureAudio(ones);
+    } else {
+      echo_detector.AnalyzeRenderAudio(zeros);
+      echo_detector.AnalyzeCaptureAudio(zeros);
+    }
+    if (i % 100 == 0) {
+      // This is causing the simulated clock drift.
+      echo_detector.AnalyzeRenderAudio(zeros);
+    }
+  }
+  // We expect to detect echo with high likelihood. Clock drift is harder to
+  // correct on the render side than on the capture side. This is due to the
+  // render buffer, clock drift can only be discovered after a certain delay.
+  // A growing buffer can be caused by jitter or clock drift and it's not
+  // possible to make this decision right away. For this reason we only expect
+  // an echo likelihood of 80% in this test.
+  EXPECT_GT(echo_detector.echo_likelihood(), 0.8f);
+}
+
+TEST(ResidualEchoDetectorTests, EchoWithCaptureClockDrift) {
+  ResidualEchoDetector echo_detector;
+  std::vector<float> ones(160, 1.f);
+  std::vector<float> zeros(160, 0.f);
+
+  // In this test the capture signal has a delay of 10 frames w.r.t. the render
+  // signal, but is otherwise identical. Both signals are periodic with a 20
+  // frame interval. There is a simulated clock drift of 1% in this test, with
+  // the capture side producing data slightly faster.
+  for (int i = 0; i < 1000; i++) {
+    if (i % 20 == 0) {
+      echo_detector.AnalyzeRenderAudio(ones);
+      echo_detector.AnalyzeCaptureAudio(zeros);
+    } else if (i % 20 == 10) {
+      echo_detector.AnalyzeRenderAudio(zeros);
+      echo_detector.AnalyzeCaptureAudio(ones);
+    } else {
+      echo_detector.AnalyzeRenderAudio(zeros);
+      echo_detector.AnalyzeCaptureAudio(zeros);
+    }
+    if (i % 100 == 0) {
+      // This is causing the simulated clock drift.
+      echo_detector.AnalyzeCaptureAudio(zeros);
+    }
+  }
+  // We expect to detect echo with near certain likelihood.
+  EXPECT_NEAR(1.f, echo_detector.echo_likelihood(), 0.01f);
+}
+
+}  // namespace webrtc